A PBX provides interconnections among internal telephone lines that are connected to telephone instruments at a single facility (such as a law office). The PBX also interconnects the internal telephone lines to a smaller number of external telephone lines (also called “trunks”) of a telephone company. Such PBXs provide a number of features of the type described in, for example, “DEFINITY Communications System, Generic 1 and Generic 3 and System 75, 8410 Voice Terminal User's Guide,” pages 5–8, 1994, published by AT&T GBCS Documentation Development, Middletown, N.J., 07748-1998.
There is a growing trend towards audio communications taking place over packet-switched networks, such as the Internet, instead of directly on the telephone network (that provides only circuit switching and is sometimes called “public switched telephone network(PSTN)”). Such audio communications can be facilitated by various types of devices such as: (1) specialized PBXs (also called “packet-switching PBXs”) that directly connect to packet-switched networks, (2) gateways that connect circuit-switching devices to packet-switched networks and (3) software tools that connect personal computers to packet-switched networks. One example of a packet-switching PBX (FIG. 1) is described in “Intranet and IP-Based UnPBXs,” Chapter 7, pages 7–16 to 7–22, in the book entitled “the UnPBX” edited by Edwin Margulies, Flatiron Publishing, Inc. 1997.
In this example, the packet-switching PBX includes one or more telephony switches 1 and 2, each of which has twelve ports that can be connected either to internal telephone lines or to external telephone lines. In addition, each of telephony switches 1 and 2 includes a digital port that is connected to an ethernet 3 for communication therebetween. For example, if telephone instrument 4 needs to be connected to telephone instrument 5, switch 2 routes the call via ethernet 3 to switch 1. Information carried by any call routed over ethernet 3 is chopped up into a number of portions, and each portion is placed in a packet (such as a UDP packet that conforms to the TCP/IP protocol used over the Internet) that is transmitted between switches 1 and 2.
One example of a gateway for circuit-switching PBXs is the ITS-E described in “Products Services & Solutions Internet Telephony Server-E”, published November 1997 and available on the Internet at http://www.lucent.com /enterprise/internet/its-e/how—its—works.html. The ITS-E includes a PSTN interface board 11 (FIG. 2) for connection to telephone lines (T1/E1/analog) of a PBX, and a DSP card 12 that performs voice compression and/or fax processing and generates packets, and the packets are sent to an ethernet 13 via an ethernet card 14.
One example of a software tool for use in a personal computer is an audio conferencing tool described in “vat—LBNL Audio Conferencing Tool”, published May 1996 and available at http://www-nrg.ee.lbl.gov/vat. The packets generated by this tool conform to the real-time transport protocol (RTP) as described in “RTP: A Transport Protocol for Real-Time Applications”, Network Working Group, January 1996, available from http://www.ietf.org/rfc/rfcl889.txt. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services.
Audio communications over packet switched networks have several drawbacks related to quality of service. For example, packet delay (corresponding to the time difference from when a first user begins talking to when a second user hears the first syllable) affects quality, as described in a paper entitled “Impact and Performance of Lucent's Internet Telephony Server (ITS) over IP Networks”, November 1997 available from http://www.lucent.com. Packet delay is of two types: a fixed delay that arises from signal processing and propagation and variable delay that results from queuing and processing of the packets.
Moreover, for good quality, the packets must be reassembled in an ordered stream and played out at regular intervals despite varying arrival times. Variation in the arrival of packets results in an effect known as “jitter.” Jitter can be handled via a buffer delay that corresponds to the maximum variable delay that is expected.
Another factor that affects service quality is loss of packets. Packet loss can be defined as the percentage of transmission packets from a source audio terminal that do not reach the destination audio terminal. Packet loss can occur due to a number of reasons, including excessive delay and congestion. Excessive delay leads to packet loss if the delay experienced by a packet exceeds the “time-to-live” value of the packet. Moreover, when queues in a router (between the source audio terminal and the destination audio terminal) grow large, the load in the router's central processing unit (CPU) increases. As the queues fill up and congestion increases, a common decongestion method is to drop all the packets in all the queues. Since audio is normally transmitted using UDP packets (that are not retransmitted), such packet loss is perceived as gaps in conversation.
When network performance deteriorates beyond a particular threshold, the above-described paper “Impact and Performance of Lucent's Internet Telephony Server (ITS) over IP Networks” recommends that calls over the packet-switched network be “blocked” or optionally routed over the regular PSTN.